| 00:04.41 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 00:05.13 | snadge | im gonna get melvined for this.. but we've inherited a whacky system from another company.. and theres no obvious place where DIDs are mapped to usernames |
| 00:05.21 | snadge | so im having a look at the internal asterisk DB |
| 00:05.39 | snadge | just a conversion to a text file |
| 00:09.27 | snadge | ok it seems theres a table, for want of a better word.. column, key name.. whatever.. called DID.. which would appear to map DIDs to accounts.. this is gonna be painful |
| 00:09.44 | snadge | if their admin system didn't actually suck balls, we would be able to just pull out a list of dids.. and copy them into our system.. sigh :p |
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| 09:22.47 | skrusty | morning |
| 09:25.28 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 09:27.22 | ChannelZ | HEY HEY HEY! |
| 09:27.41 | skrusty | :) |
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| 09:35.18 | eirirs | torning |
| 09:43.59 | *** join/#asterisk Tommy007 (~chatzilla@85.118.159.250) |
| 09:46.49 | Tommy007 | Hi I am having problems detecting DTMF for the following call scenario -SIP Call A into Asterisk into AGI, SIP Call B outdial via AMI outdial, B call answered back in via AGI, bridge A+B calls |
| 09:47.15 | Tommy007 | I've confirmed that the SIP signalling and rtp media are going via the Asterisk |
| 09:47.27 | *** join/#asterisk oatha (~oatha@201.47.215.172.dynamic.adsl.gvt.net.br) |
| 09:47.54 | Tommy007 | Also DTMF is detected fine on the console for the A-Asterisk leg prior to making the bridged outdial to B |
| 09:52.56 | *** join/#asterisk eject_ck (~eject_ck@62.205.157.98) |
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| 09:53.44 | eject_ck | Hi all, on ubuntu I've installed asterisk from packages. How can I make sure it have srtp support or not ? With ldd I see no srtp libs linked. |
| 09:56.25 | eject_ck | how I can make sure rtp is secured? |
| 09:56.35 | eject_ck | with wireshark I see it is not :( |
| 09:56.43 | eject_ck | when SIP traffic is using tls |
| 10:03.54 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
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| 11:09.23 | skrusty | Tommy007: only think i can think is maybe DTMF signalling is different in your bridged call? |
| 11:13.29 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
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| 11:23.03 | *** join/#asterisk uRDeSiRe (~yes@unaffiliated/urdesire) |
| 11:23.21 | uRDeSiRe | hello does anyone know how i can add a skype account to asterisk without using their paid SIP services? |
| 11:24.13 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.163) |
| 11:24.33 | *** join/#asterisk Ice_Strike (~Ice_Strik@cpc1-oldh7-0-0-cust772.10-1.cable.virginm.net) |
| 11:25.25 | Ice_Strike | The agents are hearing crackly static sounds sometime on the phone during outboud calls. |
| 11:25.31 | Ice_Strike | What is causing this? |
| 11:26.46 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.163) |
| 11:27.28 | whizzi | Ice_Strike: the phones. Static sound is something analog. |
| 11:28.30 | Ice_Strike | We are using sip provider. |
| 11:36.14 | vassilux | hi alls, I try send an udp packet from my asterisk module to a server, but the server never get it. I tried my code into a test program it works well. Any idea ? |
| 11:38.00 | *** join/#asterisk calum_ (~calum_@host86-128-103-152.range86-128.btcentralplus.com) |
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| 11:57.22 | Greenlight | Is there any way to use say a float or decimal for the "billsec" field of the CDR's generated? |
| 12:02.50 | whizzi | Ice_Strike: still, it's the phones or something not digital ;) |
| 12:10.16 | wdoekes | vassilux: that could be anything, you should probably show the code |
| 12:10.54 | wdoekes | or just try tracing what succeeds and what doesn't |
| 12:15.16 | vassilux | yes you right, wdoekes. |
| 12:32.20 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 12:41.54 | Ice_Strike | for 10 computers - with 30 power plugs :( |
| 12:42.03 | Ice_Strike | Monitor, Computers and Phone |
| 12:42.05 | *** join/#asterisk felipealmeida (~user@179.210.237.70) |
| 12:42.06 | Ice_Strike | x 30. |
| 12:42.12 | Ice_Strike | Err x 10. |
| 12:42.13 | wdoekes | Greenlight: hrtime=yes in cdr_odbc.conf |
| 12:42.57 | Ice_Strike | Using a few Surge Protected Power Extension.. from Extension to Extension. |
| 12:43.50 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 12:43.56 | whizzi | Ice_Strike: is it on all phones? |
| 12:44.12 | Ice_Strike | Yes |
| 12:44.17 | Ice_Strike | Happen to all the phones. |
| 12:44.39 | whizzi | to all numbers or to a specific number? |
| 12:44.51 | Ice_Strike | to any numbers yes |
| 12:45.29 | whizzi | all phones are exactly the same? Same brand, nr ? |
| 12:45.49 | Ice_Strike | Yes all actually the same. |
| 12:46.25 | whizzi | have you tried other brands/phones or even a soft client? |
| 12:52.45 | Greenlight | wdoekes: Many thanks |
| 12:52.50 | Greenlight | Ice_Strike: Bad earth? |
| 12:53.04 | Greenlight | Are you using PoE ? |
| 12:54.30 | War_Bear | Hi all, does anyone know if its possible to have asterisk use different contexts based on the calls coming from the same IP but with different domains in the to field? |
| 12:58.13 | *** join/#asterisk jploh (~textual@49.144.21.129) |
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| 13:07.32 | *** part/#asterisk eject_ck (~eject_ck@62.205.157.98) |
| 13:10.08 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 13:10.47 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:13.45 | *** join/#asterisk LooserOuting (~LooserOut@ip-109-91-207-144.unitymediagroup.de) |
| 13:15.58 | LooserOuting | Hi. I need asterisk to sent RTP as soon it receives a seesion progress so the fire will open the ports for receiving the ringing(with RTP of course, not Response 180). |
| 13:16.27 | LooserOuting | I ment firewall, not fire will :-D |
| 13:18.40 | mirela666 | fire@wil |
| 13:18.54 | LooserOuting | 0.o |
| 13:19.22 | Greenlight | Progress would do that wouldn't it |
| 13:19.47 | Greenlight | "core show application progress" |
| 13:20.01 | LooserOuting | Progress will send a Progress to the othe leg |
| 13:20.12 | Greenlight | Isn't that what you wanted ? |
| 13:20.29 | LooserOuting | no |
| 13:21.29 | LooserOuting | My Firewall ist blocking RTP because the connection is not established. |
| 13:22.09 | LooserOuting | or biderectional |
| 13:22.17 | LooserOuting | or one-way |
| 13:23.23 | LooserOuting | dunno if i am explaining it right |
| 13:23.38 | [TK]D-Fender | RTP ports should simply be forwarded directly to your server... not firewalled off. |
| 13:23.56 | Greenlight | Are you also behind NAT ? |
| 13:24.13 | LooserOuting | I am not behind NAT |
| 13:24.20 | Greenlight | So why are you firewalling the RTP ports... |
| 13:24.27 | Greenlight | And expecting it to work |
| 13:24.50 | LooserOuting | So nobody sends Data to that ports. |
| 13:24.51 | Greenlight | Your server has a public IP, so just unlock the RTP range |
| 13:24.57 | LooserOuting | no |
| 13:25.09 | [TK]D-Fender | And if Asterisk receives session progress, then that means it is the side originating the call... Asterisk should be arbitrarily deciding to INITIATE audio when the other side isn't asking for it |
| 13:25.56 | Greenlight | [01:24pm] <LooserOuting> So nobody sends Data to that ports. <-- Yes this is what you've done... but sounds like you *want* data to come in since you want the RTP traffic... |
| 13:26.15 | LooserOuting | [TK]D-Fender: there are devics that do send silence in such a case. |
| 13:26.20 | *** join/#asterisk mjordan (~matt@75.76.55.191) |
| 13:26.20 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:26.48 | LooserOuting | Greenlight: really ??? sure I want data. But only the right one. |
| 13:26.51 | [TK]D-Fender | LooserOuting: You are describing the REVERSE of early media |
| 13:27.05 | LooserOuting | [TK]D-Fender: I know :) |
| 13:27.12 | [TK]D-Fender | LooserOuting: Early media is something a receiver does. |
| 13:27.27 | Greenlight | If you want RTP then unlock those ports.. |
| 13:27.33 | Greenlight | *unblock |
| 13:27.42 | LooserOuting | I JSUT CAN'T OPEN THE PORTS FOR ALL |
| 13:27.43 | [TK]D-Fender | LooserOuting: And your request seems non-sensible and paranoid... |
| 13:27.59 | [TK]D-Fender | [08:27]LooserOutingI JSUT CAN'T OPEN THE PORTS FOR ALL <- Why Not. Everyone else does. |
| 13:28.05 | Greenlight | Well open them for the IPs you want to receive RTP from |
| 13:28.06 | [TK]D-Fender | LooserOuting: that is how this works. |
| 13:28.41 | LooserOuting | [TK]D-Fender: I know how it work. |
| 13:29.00 | [TK]D-Fender | LooserOuting: There is no option to do what you want. Deal with that reality |
| 13:29.58 | LooserOuting | [TK]D-Fender: So the reality is asterisk doesn't support that. Others do ! |
| 13:30.11 | LooserOuting | [TK]D-Fender: Thats's fine |
| 13:30.16 | *** join/#asterisk felipealmeida (~user@179.210.237.70) |
| 13:30.17 | LooserOuting | [TK]D-Fender: i am just asking |
| 13:30.23 | Greenlight | The reality is it's a silly idea and provides no benefit |
| 13:30.44 | Greenlight | How would your idea work if both ends implemeneted your idea ? |
| 13:31.21 | LooserOuting | Greenlight: Sure it provides benefits. just make a call and see what happens when other send you some pakets at that port |
| 13:31.37 | Greenlight | Umm Asterisk ignores them... |
| 13:32.11 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 13:33.25 | LooserOuting | Greenlight: So there is no potentional risk of a segfault or a DDOS for example ? |
| 13:33.32 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:33.33 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:33.41 | [TK]D-Fender | see: Paranoia |
| 13:33.45 | Greenlight | DDOS isn't mitigated here at all using your suggestion. |
| 13:34.05 | Greenlight | If you have 100mb bandwidth, and I flood it, you're scrweed either way |
| 13:34.18 | Greenlight | Wether your obscure firewall rules drops the packet, or asterisk does. |
| 13:34.44 | LooserOuting | [TK]D-Fender: it's not paranoia. It's called Quality Assurance ! |
| 13:35.30 | LooserOuting | Greenlight: My Firewall handles more then 100mb |
| 13:35.52 | Greenlight | Well whatever your internet link is. |
| 13:36.24 | [TK]D-Fender | Well it does block that from hitting the inside.. the OUTSIDE will still get flooded... |
| 13:36.54 | LooserOuting | Greenlight: The asterisk is behind that firewall so nothing will happen to the asterisk |
| 13:37.12 | LooserOuting | right |
| 13:37.21 | [TK]D-Fender | LooserOuting: Asterisk doesn't do this so go look for a proxy that will. |
| 13:37.21 | Greenlight | Apart from not being able to make or receive calls, since your bandwidth is flooded. |
| 13:37.44 | [TK]D-Fender | Greenlight: Well... at least the machine won;t be flooded on the inside |
| 13:37.56 | Greenlight | How is that helpful ? |
| 13:38.08 | [TK]D-Fender | Greenlight: So his local devices can still communicate |
| 13:38.11 | [TK]D-Fender | ^ |
| 13:38.24 | [TK]D-Fender | And perhaps a failover link can salvage things |
| 13:38.34 | *** join/#asterisk felipealmeida (~user@179.210.237.70) |
| 13:38.40 | [TK]D-Fender | there is that... but his primary link will still be hammered |
| 13:38.47 | Greenlight | I would expect the bottlneck to be bandiwdth so internal would still work fine. |
| 13:39.06 | Greenlight | I can't see a 100mb flood killing asterisk's call processing ability. |
| 13:39.22 | [TK]D-Fender | Unless it's on a 100bt NIC |
| 13:39.30 | Greenlight | Well, that's true |
| 13:42.38 | Greenlight | I guess somelink like OpenSIPs gives you more granual control over what is sent and when |
| 13:42.43 | Greenlight | *something |
| 13:42.58 | LooserOuting | Well, my qustion was anwered. So thank you for this discussion. |
| 13:45.08 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 13:51.00 | lnb | anyone know workaround for: STARTING ASTERISK |
| 13:51.02 | lnb | /usr/sbin/safe_asterisk: 90: ulimit: error setting limit (Operation not permitted) |
| 13:51.02 | lnb | Asterisk ended with exit status 127 |
| 13:51.02 | lnb | Asterisk died with code 127. |
| 13:51.02 | lnb | cat: /var/run/asterisk/asterisk.pid: No such file or directory |
| 13:51.41 | lnb | added asterisk hard nofile 3072 to /etc/secuity/limits.conf but that doesnt help either |
| 13:55.05 | whizzi | set your ulimit for the asterisk-user |
| 13:55.10 | *** join/#asterisk serafie (~erin@nat/digium/x-iwkudhrvluqwotaw) |
| 13:55.42 | whizzi | oh wait, is this by any chance a virtual system ? |
| 13:56.35 | *** join/#asterisk mjordan (~matt@nat/digium/x-lshaoiajydxbdryn) |
| 13:56.36 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:57.34 | lnb | whizzi: yes, kvm VM |
| 13:58.23 | whizzi | the set the ulimit to unlimited at the KVM-server |
| 13:58.28 | whizzi | *then |
| 13:58.42 | [TK]D-Fender | [08:57]===Unknown CTCP userinfo () from LooserOuting |
| 13:58.52 | [TK]D-Fender | LooserOuting: Scanning is not a great way of making friends... |
| 13:59.28 | lnb | whizzi: where? |
| 13:59.43 | lnb | you mean on the host server? |
| 14:00.01 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 14:00.16 | Blashyrkh | what does the n+x stand for in an asterisk dialplan |
| 14:00.18 | LooserOuting | [TK]D-Fender: just being curious. sorry for that. |
| 14:00.20 | Blashyrkh | where x is a number |
| 14:00.30 | whizzi | lnb: aye |
| 14:00.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.163) |
| 14:02.15 | lnb | whizzi: vim /etc/security/limits.conf ? |
| 14:05.22 | [TK]D-Fender | Blashyrkh: Show us your sample |
| 14:05.42 | [TK]D-Fender | Blashyrkh: As that should be "current+more" |
| 14:06.05 | [TK]D-Fender | Blashyrkh: It is NOT actual syntax, but rather just the math of the priority number |
| 14:06.18 | [TK]D-Fender | Blashyrkh: You should never have a "+" in the dialplan. |
| 14:06.21 | *** join/#asterisk leifmadsen (~free-mads@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:06.21 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:06.32 | [TK]D-Fender | (in the priority that is |
| 14:07.14 | Blashyrkh | iam just looking through a dialplan and there are a lot of things like this |
| 14:07.14 | Blashyrkh | exten => _X.,n+1(no_premium),Goto(${CUT(inbound_info,",",2)},${EXTEN},1) |
| 14:07.14 | Blashyrkh | exten => _X.,n,Hangup() |
| 14:07.29 | Blashyrkh | and iam a bit confused about the n+1() |
| 14:07.38 | [TK]D-Fender | doesn't look legit |
| 14:08.10 | kaldemar | it should be legit but there's seldom any use for that. |
| 14:08.31 | kaldemar | Blashyrkh: what's in the rest of the extension matters. |
| 14:08.36 | Blashyrkh | i found it here too |
| 14:08.37 | Blashyrkh | http://www.voip-info.org/wiki/view/Asterisk+priorities |
| 14:08.44 | whizzi | lnb: yes :) |
| 14:08.54 | Blashyrkh | i just wanted to no what this is for in general |
| 14:09.00 | Blashyrkh | or what it does |
| 14:09.12 | Blashyrkh | without the +1 its just a label |
| 14:09.29 | lnb | whizzi: the username would be what output of virsh list shows ? |
| 14:09.38 | lnb | for the VM ? |
| 14:09.46 | whizzi | it's best to set everything to unlimited |
| 14:09.56 | kaldemar | Blashyrkh: was this not explained yesterday already? |
| 14:09.58 | [TK]D-Fender | kaldemar: That looks.... so very wrong.. that page refers to is with regards to priority jumping which is LONG since dead... |
| 14:10.07 | lnb | what should the line look like? |
| 14:10.10 | [TK]D-Fender | kaldemar: What the hell is +1 in this case? |
| 14:10.24 | kaldemar | [TK]D-Fender: that's what i said yesterday. |
| 14:11.08 | [TK]D-Fender | Blashyrkh: that WIKI is full of ANCIENT garbage |
| 14:11.57 | kaldemar | [TK]D-Fender: n+1 is really n+1. if you have priorities 1, n and n+1, those are parsed to 1, 2 and 4. |
| 14:12.30 | [TK]D-Fender | kaldemar: I'm trying to imagine the POINT. * at least used to have a hard-coded +101 for Dial, etc |
| 14:12.46 | [TK]D-Fender | kaldemar: Arbtrary numbers is looking REAL dumb now... |
| 14:12.57 | kaldemar | [TK]D-Fender: don't bother, there is no point whatsoever in that anymore. |
| 14:12.58 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:13.12 | [TK]D-Fender | kaldemar: Forget now.. I can't even see a point THEN :p |
| 14:13.50 | kaldemar | [TK]D-Fender: maybe in the 1.2 era there was a point for someone who insisted in sticking with priority jumping AND n-priorities. |
| 14:15.46 | [TK]D-Fender | kaldemar: I'm wonding what would target it as to why you'd skip. The +101 was at least a known relative target and you could predict the dialplan app of the line before that would warrant it. Where would you find a point to an arbitrary relative jump? |
| 14:15.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.170) |
| 14:16.50 | [TK]D-Fender | kaldemar: Not withstanding that in order to do that... you couldn't have n's for if it DIDN'T fail.. it just auto-bumps the sequence. This is just so full of fail.... |
| 14:21.28 | kaldemar | [TK]D-Fender: tickets to fail train from voip-info. |
| 14:22.09 | [TK]D-Fender | kaldemar: well... the fact it's dated doesn't meant he concept wasnt true... and pointless even then... |
| 14:22.11 | *** join/#asterisk defswork (~andy@141.0.50.105) |
| 14:28.06 | *** part/#asterisk mjordan (~matt@nat/digium/x-lshaoiajydxbdryn) |
| 14:29.59 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 14:30.09 | *** join/#asterisk theron (~theron@69.63.185.56) |
| 14:33.55 | lnb | trying to compile/install asterisk from tar.gz, after installing, asterisk -r : error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory |
| 14:34.26 | lnb | what is supposed to install the missing ssl.so.1 ? |
| 14:36.17 | [TK]D-Fender | Quite certainly missing required libraries.... |
| 14:36.28 | [TK]D-Fender | Which menuselect would have poitned out.. |
| 14:38.10 | lnb | menuselect allowed me to select what i needed.. res_mysql and mysql_cdr |
| 14:38.57 | lnb | going to run make uninstall-all && make distclean && make clean |
| 14:39.17 | lnb | i tried adding a few more pkg installs perhaps that will help |
| 14:40.21 | lnb | if i run ./configure with no args, i presume it will install /usr/lib and since its 64bit os, it should be /usr/lib64 |
| 14:40.27 | [TK]D-Fender | <PROTECTED> |
| 14:40.37 | [TK]D-Fender | Which is what everyone shold be using. |
| 14:41.13 | lnb | while you might be 100% right, the page I am using to help install says to install anyway, along with unixodbc et al |
| 14:41.34 | lnb | http://www.powerpbx.org/content/asterisk-freepbx-install-guide-ubuntu-v12-asterisk-v11-freepbx-v211 |
| 14:41.48 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 14:42.06 | lnb | but so far, when trying to run ./start_asterisk start it fails |
| 14:42.17 | lnb | from freepbx install dir |
| 14:42.57 | [TK]D-Fender | They even say that it is an older method and isn't what they advise first |
| 14:43.20 | [TK]D-Fender | Install CDR ODBC required packages: This is required if you plan to use the new default behaviour which is to use the asterisk module res_config_odbc to connect to CDR DB. If you want to use the old simpler method make sure to select the asterisk cdr_mysql module at compile time from menuselect. If unsure, install the following packages anyways. |
| 14:43.34 | tparcina | I have safe_asterisk script on one Debian computer, but it doesn't run when I power on the computer. |
| 14:43.51 | tparcina | What is the prefered way to start it? |
| 14:44.01 | [TK]D-Fender | Also, you are following some random guide and not official * docs |
| 14:44.13 | lnb | i did install odbc packages |
| 14:44.20 | [TK]D-Fender | tparcina: standard init script... |
| 14:44.37 | [TK]D-Fender | [09:38]lnbmenuselect allowed me to select what i needed.. res_mysql and mysql_cdr <- then you shouldn't be using these at all |
| 14:44.42 | lnb | apt-get install -y unixodbc unixodbc-dev libmyodbc |
| 14:44.45 | tparcina | [TK]D-Fender: OK, thank you. |
| 14:44.59 | lnb | why |
| 14:45.05 | lnb | need realtime |
| 14:45.14 | lnb | as this will be an a2billing server |
| 14:45.50 | [TK]D-Fender | Realtime does not requre direct MySQL |
| 14:46.01 | lnb | ok |
| 14:46.21 | lnb | i am totally lost now |
| 14:46.27 | lnb | whats new |
| 14:46.41 | [TK]D-Fender | Because you are always doing everything EXCEPT learning Asterisk |
| 14:47.20 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:47.20 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:47.30 | lnb | ok this time not going to check mysql_cdr |
| 14:47.34 | lnb | in menuselect |
| 14:47.40 | lnb | just res_mysql |
| 14:48.52 | [TK]D-Fender | And what requires that? |
| 14:49.03 | lnb | since this box is only for a2b, then i assume there is no logic to adding sounds beyone the default in menuselect |
| 14:49.28 | lnb | FreePBX does not use Asterisk realtime but if you are thinking of using A2Billing then also select res_config_mysql |
| 14:49.54 | [TK]D-Fender | ....because? |
| 14:50.01 | [TK]D-Fender | What actual FUNCTION is using it? |
| 14:50.08 | lnb | i have NO CLUE |
| 14:50.18 | lnb | only that it says to select it |
| 14:50.35 | [TK]D-Fender | So in addition to not having learned Asterisk .. you haven't leaned A2B either |
| 14:50.37 | lnb | and even Joe @ star2billing says its used |
| 14:54.31 | *** join/#asterisk mjordan (~matt@nat/digium/x-vdqriuhhtaenhmci) |
| 14:54.31 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:57.34 | lnb | does [*] res_config_mysql have to be installed, i said because its needed for a2billing |
| 14:57.34 | lnb | Star2Billing Sales: yes, thats correct. |
| 14:57.57 | [TK]D-Fender | Doesn't answer how it's supposedly being used |
| 14:58.13 | [TK]D-Fender | because that could be on an assumption as to how the backend work gets done |
| 14:58.27 | [TK]D-Fender | This is the kind of thing that can happen when asked shitty questions |
| 14:58.33 | tparcina | [TK]D-Fender: I should make another init.d script that will start safe_asterisk, or I should use safe_asterisk as init.d script? |
| 14:58.38 | lnb | Star2Billing Sales: Asterisk realtime |
| 14:58.49 | [TK]D-Fender | REALTIME... does not require re_mysql |
| 14:59.08 | [TK]D-Fender | realtime is how * pulls it.. doesn't say what the BACKEND access modeul is |
| 14:59.38 | [TK]D-Fender | tparcina: * comes with samples for RH, debian, etc |
| 14:59.47 | [TK]D-Fender | tparcina: make config |
| 15:00.18 | tparcina | [TK]D-Fender: OK, thank you. |
| 15:01.15 | [TK]D-Fender | tparcina: I'd recommend using the base script, and tweaking it if your needs vary from the base |
| 15:01.36 | lnb | Star2Billing Sales: What you are asking here is getting very close to providing you with installation support. |
| 15:01.37 | lnb | The SIP peers are contained in a table in MySQL database, which asterisk realtime queries. |
| 15:01.37 | lnb | sorry, this guy is pretty smart and he is saying it |
| 15:03.00 | lnb | Star2Billing Sales: The logs will tell you why, |
| 15:03.01 | lnb | meanwhile, why dont you use my Asterisk / FreePBX installation script. |
| 15:03.01 | lnb | INSTALL-BALLISTIC-PBX-NOTES |
| 15:03.01 | lnb | https://github.com/jonathan-roper/Ballistic-PBX/blob/master/INSTALL-BALLISTIC-PBX-NOTES |
| 15:03.57 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:03.57 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:04.30 | [TK]D-Fender | [10:01]lnbThe SIP peers are contained in a table in MySQL database, which asterisk realtime queries. <- and realtime can use ODBC to get to MySQL. |
| 15:04.46 | [TK]D-Fender | lnb: While you may think he's smart, you can't put 2 and 2 together. |
| 15:04.59 | [TK]D-Fender | lnb: that is NOT something that states a requirement |
| 15:05.06 | lnb | i said YOU'RE smart dammit |
| 15:05.50 | [TK]D-Fender | if that's copy/pasted chat back from elsewhere it wasn't clear that it was so. |
| 15:05.56 | lnb | you get a 'high' putting people down? |
| 15:06.10 | [TK]D-Fender | if that's copy/pasted chat back from elsewhere it wasn't clear that it was so. <- |
| 15:06.10 | lnb | i get a high on something else |
| 15:06.15 | wdoekes | lnb: D-Fender isn't the only one who cannot see who is who when you speak |
| 15:06.29 | lnb | he is free to talk to Joe |
| 15:06.36 | lnb | http://www.star2billing.com/ |
| 15:06.45 | [TK]D-Fender | [10:01]lnbsorry, this guy is pretty smart and he is saying it <- this does not have my nick in it. This wording means talking about someone else. And I don't see someone else's name as the target] |
| 15:06.46 | wdoekes | I have no idea when you paste stuff vs. when you write stuff in here either |
| 15:06.54 | lnb | Joe is not on iRC as far as i know |
| 15:07.13 | lnb | from joe: Star2Billing Sales: yes thats another way of doing it. |
| 15:07.16 | [TK]D-Fender | yes well you posed it without us knowing that line was from him. |
| 15:07.26 | whizzi | lnb: do you have a particular reason to compile Asterisk from scratch while you seem to have a Debian system? |
| 15:07.30 | lnb | i did not hid it |
| 15:07.32 | lnb | hide |
| 15:07.36 | lnb | look carfully |
| 15:07.40 | whizzi | lnb: I mean, apt-get install asterisk |
| 15:07.54 | lnb | <lnb> Star2Billing Sales: |
| 15:08.02 | WIMPy | Is this trying to get competing installation support? |
| 15:08.13 | lnb | whizzi: yes, I was told to |
| 15:08.28 | lnb | and i cannot afford roughly 800 Euros right now |
| 15:08.35 | lnb | to get this all done |
| 15:08.38 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 15:09.05 | lnb | morning navaismo |
| 15:09.27 | navaismo | morning |
| 15:09.37 | lnb | i need to pay to get expertise on how to effectly run a2billing |
| 15:10.01 | lnb | if not, just tell everyone ... hey call the world,,,, its FREE,... its my dime |
| 15:10.11 | navaismo | uh? |
| 15:10.17 | blitzrage | settle down kids |
| 15:10.21 | lnb | you missed the argument navaismo |
| 15:10.39 | [TK]D-Fender | argv[-1] |
| 15:17.08 | lnb | what i really like about a2b even without allowing intnl calls, if there needs to be a trunk change, simply tell that to a2b and all the freepbx or freeswitch server now use the new trunk instantly. |
| 15:17.42 | lnb | beats going to each and every fpbx/freeswitch server and doing it manually |
| 15:18.37 | [TK]D-Fender | doesn't really apply. Your other servers go through this main A2B for routing. To them YOU are the central and the fact you change your route for the next leg is never their concern |
| 15:18.54 | [TK]D-Fender | And has nothing to do with being A2B... it's for the fact it's going through your server all the time anyway |
| 15:19.01 | [TK]D-Fender | any change you do is transparent tho them |
| 15:19.07 | [TK]D-Fender | A2B isn't the magic that provides it |
| 15:19.10 | lnb | it does apply |
| 15:19.19 | lnb | yes it is |
| 15:19.21 | lnb | the magic |
| 15:19.48 | lnb | all freepbx / freeswitch boxes have one gateway/trunk <-> a2b server |
| 15:20.10 | [TK]D-Fender | No, the fact you don't have to mod those other servers has nothing to do with A2B. It the fact its your server doing the work. the fact that A2B makes THAT one esasy is besides the poitn |
| 15:20.19 | lnb | there is no trunk/gateway to any itsp on any of the fbpx servers installed |
| 15:20.30 | [TK]D-Fender | there is.... one to YOU |
| 15:20.32 | [TK]D-Fender | ^ |
| 15:20.50 | lnb | what????? |
| 15:20.57 | lnb | oh man |
| 15:21.03 | [TK]D-Fender | they have a rtunk to your A2B server |
| 15:21.16 | lnb | i need a demerol |
| 15:21.50 | lnb | yes, every fbpx and freeswitch server -> a2b server -> itsp |
| 15:21.57 | lnb | and back |
| 15:22.14 | lnb | there was an issue with one itsp a month or two ago |
| 15:22.31 | [TK]D-Fender | Yes well A2B isn't what saves having to make changes on all those other servers. |
| 15:22.52 | [TK]D-Fender | that just makes configuring your intermediary server easy for YOU |
| 15:22.52 | lnb | instead of a whole bunch of people calling that they cannot dial out iwth sip phones, make one change in a2b, problem gone |
| 15:23.02 | [TK]D-Fender | You could just as well have configured it by hand YOURSELF |
| 15:23.09 | lnb | i dont want pissed off clients |
| 15:23.15 | lnb | that does make sense doesnt it? |
| 15:23.18 | [TK]D-Fender | A2B is crap to make YOUR life easy for not having to learn anything |
| 15:23.47 | lnb | i will take the time to learn it when I can get done what I need to get done |
| 15:23.56 | [TK]D-Fender | You are mis-attributing credit |
| 15:24.00 | lnb | i dont have time now to sit and learn |
| 15:24.55 | [TK]D-Fender | I know how long you've been at this.... i'm not holding my breath for it to ever happen. |
| 15:25.10 | lnb | STARTING ASTERISK |
| 15:25.11 | lnb | /usr/sbin/safe_asterisk: 90: ulimit: error setting limit (Operation not permitted) |
| 15:25.11 | lnb | Asterisk ended with exit status 127 |
| 15:25.16 | lnb | wtf! |
| 15:25.28 | lnb | i will try Joe's script |
| 15:25.32 | lnb | this is a nitemare |
| 15:26.01 | lnb | asterisk -r |
| 15:26.02 | lnb | asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory |
| 15:27.11 | navaismo | dont know what happen here but lnb have you considered to make a feature request to A2B, pay for the development or something?? Everything is possible with money. Remember with money you make me dance |
| 15:29.44 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 15:31.35 | [TK]D-Fender | navaismo: That's against his business model. |
| 15:32.58 | navaismo | LOL |
| 15:34.08 | navaismo | I guess everyone knows the case of partedMagic if we dont donate all FOSS project will cost in the nearest future :( |
| 15:35.12 | lnb | navaismo: you have setup a2b i think you had mentioned |
| 15:35.21 | lnb | or maybe it was elastix |
| 15:35.26 | lnb | which has a2b in it |
| 15:35.34 | navaismo | twice in a lifetime |
| 15:36.01 | lnb | its not so much a feature we need, its how to really setup a2b for billing |
| 15:36.18 | lnb | since its theirs it makes sense to use them |
| 15:36.30 | navaismo | fork it |
| 15:36.35 | navaismo | create new one |
| 15:36.40 | navaismo | pay for new features |
| 15:36.49 | lnb | all i am trying to do is get the base setup.. asterisk, freepbx, and a2billing installed |
| 15:37.01 | lnb | but having issues with asterisk |
| 15:37.20 | lnb | from freepbx install DIR ./start_asterisk start failes |
| 15:37.23 | lnb | fails |
| 15:37.38 | lnb | and same with asterisk -r |
| 15:38.23 | lnb | i am going to try joe's script, in meantime deleted that VM and install new VM so nothing lingers from before |
| 15:38.29 | eppigy | hello |
| 15:38.53 | [TK]D-Fender | eppigy: you are dave |
| 15:39.07 | eppigy | yes |
| 15:39.08 | eppigy | yes i am |
| 15:39.12 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 15:39.17 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 15:45.15 | *** join/#asterisk boratynskikamil (6de72695@gateway/web/freenode/ip.109.231.38.149) |
| 15:45.41 | boratynskikamil | Good afternoon. One small question. Is there any way to keep passwords in SIP.conf not as plaintext? |
| 15:45.59 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 15:48.40 | wdoekes | boratynskikamil: md5secret is the best you can do.. that'd be the ha1 part |
| 15:48.43 | wdoekes | see digest auth |
| 15:49.10 | boratynskikamil | wdoekes: Great! Thanks a lot, I will read about that! |
| 15:49.59 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 15:50.01 | boratynskikamil | wdoekes: One small question, if I might... Is it quite secure to create SIP-accouts without password? I'm sure my local network is closed. |
| 15:50.35 | mjordan | if someone has access to the system running Asterisk, the passwords in sip.conf are the least of your worries |
| 15:51.10 | boratynskikamil | mjordan: None, besides me. |
| 15:51.16 | boratynskikamil | mjordan: I'm thinking about... login to Asterisk. |
| 15:51.25 | boratynskikamil | As SIP client. |
| 15:52.06 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.130) |
| 15:55.41 | agoodm | Hello again all, spoke you all yesterday regarding troubleshooting if a jitter buffer was working or not, I think ive found the issue causing the call quality issues but I have ran the suggested test anyway |
| 15:56.23 | agoodm | made a test call to the system over the internet connection of 3 hours or so this morning, from a known stable site, captured the traffic which flowed for this call and ran it through wireshark rtp analyser |
| 15:58.08 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.170) |
| 15:58.54 | agoodm | what is skew? |
| 16:00.26 | agoodm | im seeing max jitter 25ms which looks fine, but max skew 150ms and max delta 115ms |
| 16:06.42 | *** join/#asterisk gtjoseph (~gtjoseph@unaffiliated/gtj) |
| 16:08.27 | *** join/#asterisk petris (~petris@192.184.93.147) |
| 16:13.12 | jmls | hey |
| 16:13.56 | jmls | does anyone have an update to date link (asterisk 11 / 12) for the sql database definitions for realtime queues and queuemembers ? |
| 16:17.20 | whizzi | hmm.. my asterisk suddenly stopped putting records in the CDR table |
| 16:18.53 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 16:23.27 | whizzi | ah, found it |
| 16:23.38 | whizzi | cdr table marked as crashed |
| 16:24.40 | *** join/#asterisk Ruel_ (~root@149.255.100.107) |
| 16:42.07 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.249) |
| 16:47.14 | *** part/#asterisk whizzi (~whizzi@82-171-3-8.ip.telfort.nl) |
| 16:53.53 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 16:58.22 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 16:58.40 | mjordan | jmls: 11 != 12 |
| 16:58.49 | mjordan | jmls: however, for 12, you can take a look at what alembic produces |
| 16:59.12 | mjordan | as that should be up to date. There is a known bug with the realtime extensions table produced for MySQL in Asterisk 12, but other than that, it should be correct |
| 16:59.32 | mjordan | if you have a tarball of Asterisk 12.0.0, that has the scripts produced automatically for it |
| 16:59.54 | mjordan | jmls: https://wiki.asterisk.org/wiki/display/AST/Managing+Realtime+Databases+with+Alembic |
| 17:01.21 | jmls | mjordan: but alembic does not have the table defs for 12 for queues |
| 17:01.37 | jmls | or queue_members or cdr |
| 17:01.39 | jmls | ;) |
| 17:02.06 | mjordan | hm. I guess we did miss that. |
| 17:02.13 | jmls | *grin* ;) |
| 17:02.20 | mjordan | 11 does have the queues table for postgresql |
| 17:02.31 | jmls | this is what started my voyage of discovery yesterday ;) |
| 17:02.34 | mjordan | jmls: file a bug report for that if you don't mind. If you feel confident in writing the patch, it would be appreciated :-) |
| 17:02.40 | jmls | lol |
| 17:02.58 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 17:03.37 | mjordan | but yes, in general, you should never go to voip-info. |
| 17:03.51 | mjordan | the odds of it being correct have diminished greatly over time. |
| 17:04.06 | Katty | morning |
| 17:04.10 | jmls | yeah, I see that |
| 17:04.20 | jmls | so, on http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue we have "musiconhold VARCHAR(128)" but in queues.conf we have "musicclass" ... |
| 17:05.23 | jmls | so how does that match up ? I've looked at the source, and it seems to read the column names, and match to the .conf parameter name (converting _ to - along the way) |
| 17:05.52 | jmls | or is that because I'm looking at 11 sql, with 12 .conf ? |
| 17:07.17 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 17:10.27 | *** join/#asterisk serafie (~erin@nat/digium/x-uryjlsjlwzlagguk) |
| 17:10.56 | jmls | mjordan: https://issues.asterisk.org/jira/browse/ASTERISK-23233 |
| 17:11.05 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 17:11.25 | jmls | couldn't find an appropriate category for alembic, so used contrib/general |
| 17:16.48 | mjordan | jmls: thanks |
| 17:17.03 | *** join/#asterisk pringlescan (~Adium@static-96-245-76-214.phlapa.fios.verizon.net) |
| 17:17.22 | pringlescan | hello all, in asterisk 1.8 what's the maximum number of seconds you can use for a timeout in a queue? |
| 17:20.26 | pringlescan | It looks like it may be javascript:void(0); but I can't find it in the documentation. |
| 17:22.44 | pabelanger | how long do you want somebody in the queue for? |
| 17:23.13 | pabelanger | maybe 2147483647 seconds |
| 17:23.17 | pabelanger | what ever an int is |
| 17:23.52 | pabelanger | basically, years |
| 17:39.29 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.129) |
| 17:55.02 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 17:55.08 | [TK]D-Fender | I'd like to know where he got that concept of JAVASCRIPT as a defining thing about *'s value limit... |
| 17:55.51 | pabelanger | some GUI I would imagine |
| 17:57.30 | [TK]D-Fender | Cause yeah.. that's reliable as a basis of comparison :) |
| 18:10.45 | *** join/#asterisk joshaidan (~brianj@S010698fc113e438d.tb.shawcable.net) |
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| 19:31.13 | Katty | afternoon |
| 19:32.19 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 19:33.09 | paulc | Well hello there! |
| 19:33.11 | paulc | waves |
| 19:34.12 | Katty | waves to paulc |
| 19:34.22 | Katty | hello, i would like to science as well please. is there an opening |
| 19:35.03 | paulc | Uh... "foreign job applicant"? or...? |
| 19:35.20 | Katty | https://scontent-b-ord.xx.fbcdn.net/hphotos-prn1/t1/1511652_751383871549293_1553910069_n.jpg |
| 19:35.49 | paulc | Hahaha that's awesome |
| 19:36.48 | Katty | it's quite clever. |
| 19:38.32 | paulc | Is it home time yet? I'm kinda ready.. |
| 19:39.48 | Katty | soon! |
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| 19:48.39 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 19:50.13 | carrar | Katty: yes you can leave for home now |
| 19:50.18 | carrar | Permission granted |
| 19:50.41 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:53.13 | *** join/#asterisk thenod (~thehotmus@monitor.rhyzome.ca) |
| 19:53.21 | thenod | Hello! |
| 19:53.23 | thenod | new here |
| 19:54.10 | thenod | Can anyone discuss call parking and a ZOMBIE channel |
| 19:54.36 | thenod | on a much older version of asterisk mind you 1.6.2.9 |
| 19:55.50 | Katty | WOO! |
| 19:55.52 | Katty | poofs |
| 19:56.15 | Katty | thenod: omnombrainsnomnom? |
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| 20:06.02 | jbeez | anyone know of an android sip client that can xfer calls? |
| 20:06.37 | pabelanger | bria |
| 20:06.59 | jbeez | thx, gonna check that out |
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| 20:29.28 | *** join/#asterisk rampage73 (~rampage73@74.118.119.142) |
| 20:31.34 | rampage73 | have an issue with voicemail on transfer, what I mean is call comes in and is answered then transfered to another extension, it rings then there is a beep and nothing. otherwise voice mail is fine for direct calls from to extensions. I have started searching Google but so far have not found my exact problem. |
| 20:39.38 | Penguin | I would like to see the dialplan involved as well as sip debug with verbose turned up. |
| 20:45.15 | *** part/#asterisk rampage73 (~rampage73@74.118.119.142) |
| 20:46.22 | [TK]D-Fender | and ... *poof* |
| 20:47.35 | [TK]D-Fender | He was a Trixbox user 2.5 years ago... maybe still is... |
| 20:48.07 | [TK]D-Fender | Good odds still on a GUI system if not the same one.... |
| 20:48.44 | Penguin | Maybe the request for technical information was too scary. |
| 20:49.08 | [TK]D-Fender | http://ibot.rikers.org/%23asterisk/20110922.html.gz |
| 20:49.44 | [TK]D-Fender | Hree we see him tip off what he was using (foncore), then tries to show dialplan that was fauly, but ends up showing even that was fabricated... |
| 20:50.21 | [TK]D-Fender | Long trip to being honest about what he's looking for support for. |
| 20:50.54 | [TK]D-Fender | in fact.... in the end of that... 17:45.40p3nguinrampage73: We don't support trixbox here. |
| 20:50.56 | [TK]D-Fender | lol |
| 20:51.24 | [TK]D-Fender | 17:46.47rampage73p3nguin, I will look and see if I can find it <- certainly had no clue where/how to look then |
| 20:51.37 | [TK]D-Fender | hugs infobot |
| 20:51.44 | [TK]D-Fender | ~botsnack |
| 20:51.44 | infobot | [TK]D-Fender: aw, gee |
| 20:53.49 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 21:24.01 | Kobaz | are there any plaintext authentication methods for sip? |
| 21:24.19 | Kobaz | like any way an auth can get sniffed and re-used by an attacker? |
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| 21:43.49 | amessina | sruffell: A question about the change made in dahdi-tools-2.9.0.1... the udev scripts never seem to call dahdi_cfg since auto_assign_spans=1. I have only a Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express) which used to load and configure "automatically" via udev. |
| 21:44.54 | amessina | Do I need to set auto_assign_spans=0 and create an entry in assigned-spans.conf to get it to auto-apply dahdi_cfg? |
| 21:45.00 | sruffell | I'm confused…that change was to prevent dahdi_span_config from actually trying to configure the spans when auto_assign_spans=1 (which is still the default…that didn't change in this release) |
| 21:45.35 | amessina | I think the "exit 0" means that none of the scriptlets in handle_device.d will ever run |
| 21:45.56 | sruffell | Yes…if you want to use the (still being tested) new configuration mechanism, you'll need to set auto_assign_spans=0. |
| 21:46.30 | sruffell | But if you do that, and you have all the new configuration files in place…your driver should already be completely setup when the dahdi init script runs. |
| 21:46.51 | amessina | ok, well other than they myriad of separate scripts ;) I have been able to successfully NOT use /etc/init.d/dahdi and use UDEV instead (up until 2.9.0.1) |
| 21:47.14 | sruffell | cool Yeah….we've been working to try and prevent an explosion of new scripts. |
| 21:47.21 | amessina | I'll set auto_assign_spans=0 to allow UDEV to run the handle_device.d scripts. |
| 21:47.48 | sruffell | That's awesome that you're using the new scripts. |
| 21:50.04 | amessina | I have been since 2.6 |
| 21:50.19 | amessina | This works: auto_assign_spans=0 to allow UDEV to run the handle_device.d scripts. |
| 21:50.49 | amessina | It seems strange that since I want something "automatic", I have to set auto_assign_spans=0, doesn't it? |
| 21:51.27 | sruffell | heh..I didn't think of that. but that parameter is whether dahdi.ko should automatically assign the spans, or leave them for the hotplug scripts to be assigned. |
| 21:51.56 | sruffell | so..if auto_assign_spans=0, all the hotplug scripts should assume that "legacy" configuration is being done, where driver load order determines span_number. |
| 21:52.11 | amessina | Right, but it is a little confusing. |
| 21:52.20 | sruffell | There were some commits to allow driver load order to be honored in certain situations (when the new config files are not present) but that wasn't finished in time for 2.9. |
| 21:52.24 | sruffell | Hmm... |
| 21:53.28 | amessina | Ok, but if I use auto_assign_spans=1, dadhi_cfg will never be reun by UDEV at all. How then does it run (in the proposed/future scenario)? |
| 21:53.30 | sruffell | yeah….I'm trying to get a new dahdi-dev mailing list setup (that is configured more like lkml…without the assumption that all traffic moves through it)…when/if that happens I'll try to loop you into any 2.10 discussions since you're actually using the new scripts. |
| 21:54.26 | amessina | Ok, thanks. |
| 21:54.54 | sruffell | so..by default now…auto_assign_spans=1….all the setup is done by /etc/init.d/dahdi. In the future the hotplug scripts will check if it knows about the span in /etc/dahdi/assigned-spans.conf. If it does…automatically set them up. If it doesn't, then when /etc/init.d/dahdi runs, it will assign any unknown spans in registration order and proceed. |
| 21:56.41 | amessina | I see. I'm working toward a future where there is no /etc/init.d/dahdi (since I haven't used one at all in a while) and only rely on the UDEV scripts to do the work when the module/device is detected by the kernel. |
| 21:57.07 | amessina | I suppose I should add my card to assigned-spans.conf (even though there is just the one card) |
| 21:59.15 | sruffell | I don't think /etc/init./dahdi will be able to be completely dropped for awhile, since it's needed for dynamic_spans in addition to just loading dahdi for people who are only loading it for timing or app_meetme. |
| 22:00.45 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.216) |
| 22:01.03 | amessina | sruffell: /etc/init.d/dahdi --> systemd.service or systemd.device (for Fedora) |
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| 22:01.27 | sruffell | ahh…yeah….that'll work too. |
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| 22:23.22 | amessina | sruffell: perhaps auto_assign_spans could be called udev_assign_spans and the logic reversed so that udev_assign_spans=1 would be the equivalent of what exists now (with auto_assign_spans=0), but it would be more clear |
| 22:24.34 | sruffell | hmm..I'm not sure, since what you're talking about is not the auto assignment of spans, but the automatic configuration of spans. For historical reasons those are separate. |
| 22:24.35 | amessina | I guess the end result could be that the default would be that udev assigns the spans, since that way, new users and users with only one card will have a working config from the start. |
| 22:25.19 | sruffell | that was the goal…but there was too much pushback about new things to grok. It will be the goal in the next release of DAHDI…. |
| 22:25.24 | amessina | Right now, new users or users with only one card have to set auto_assign_spans=0 to configure the card, which seems like it would be difficult for a new user to track down. |
| 22:25.33 | amessina | Ok, just thinking out loud. |
| 22:26.12 | sruffell | no..no problem. I appreciate your comments. The "standard" way for new users to configure their cards is the same: modprobe driver; dahdi_genconf; /etc/init.d/dahdi restart |
| 22:26.55 | amessina | right, I was thinking the |
| 22:27.34 | amessina | ... "standard" way of the future (no default /etc/init.d/dahdi) -- let the system/kernel/udev do what it does best--load devices and drivers |
| 22:28.42 | sruffell | I completely agree. That is the goal. Still some more work in user space to allow that to happen. I.e…how to configure the scripts to remember existing cards so that when new cards are added they don't conflict, how to allow it to work when people are just updating from a previous version of dahdi and they do not want to touch their config scripts, etc.. |
| 22:29.09 | amessina | Yup. |
| 22:29.25 | sruffell | also need to move all the module parameter configuration items (opermode, etc..) to a config file so that these can be automatically guessed based on locale…. |
| 22:30.36 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:31.24 | bsdice | Good news! My phone still works... with Asterisk! |
| 22:31.32 | bsdice | Now trying Confbridge trickery |
| 22:32.20 | amessina | sruffell: I guess there's more work to do, but generally things are working well without the old /etc/init.d/dahdi. Good work. Have a good evening and thanks for your help. |
| 22:32.56 | sruffell | have a good night. |
| 22:40.54 | *** join/#asterisk yang (yang@freenode/sponsor/fsf.member.yang) |
| 22:45.16 | bsdice | What do you guys do to protect conference channels from PIN-guessing? |
| 22:45.36 | bsdice | Background playback of modem noise to fool asshats? |
| 22:46.39 | bsdice | (while PIN must be entered) |
| 22:47.54 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
| 22:48.26 | [TK]D-Fender | What stage does this background noise come in to protect it? |
| 22:48.36 | bsdice | Greeting ;-) |
| 22:48.45 | *** join/#asterisk timeshell (~timeshell@184.151.61.168) |
| 22:49.01 | bsdice | machine answers to call, plays modem negotiation |
| 22:50.20 | bsdice | if you are a hacker, what do you do? enter 8 digit pin? or hang up |
| 22:50.32 | bsdice | because you think you have reached a fax machine ;-) |
| 22:51.17 | [TK]D-Fender | What machine answers "to call"? Why is it playing a modem negotiation? |
| 22:51.34 | bsdice | to fool outsiders |
| 22:51.43 | [TK]D-Fender | Your whole wording is very broken. Please rephrase your scenario |
| 22:51.51 | bsdice | it waits for pin while playing modem negotiation wav to caller |
| 22:52.04 | bsdice | and doesn't say "hey, thanks for calling, enter pin" |
| 22:52.39 | [TK]D-Fender | Ask for a pin with your own recording, as noisy as you want it to be..... |
| 22:53.00 | [TK]D-Fender | And still... what are they "detecting"? |
| 22:53.20 | [TK]D-Fender | Who is calling this number? |
| 22:53.32 | bsdice | attackers might call |
| 22:53.38 | bsdice | looking for attack target |
| 22:53.47 | bsdice | see its a conferencing system, try PINs |
| 22:54.01 | bsdice | don't you get it? |
| 22:54.49 | [TK]D-Fender | How is an attacking machine going to evaluate "this is a conference system asking for a pin"? |
| 22:55.08 | bsdice | it won't, this will be a human |
| 22:55.18 | [TK]D-Fender | how would it even evaulat a successful attack if it did? |
| 22:55.40 | [TK]D-Fender | Your prompt is still SAYING it's a conferencing system ,enter a pin, no? |
| 22:55.47 | bsdice | no |
| 22:55.51 | [TK]D-Fender | What does background noise doto fool a HUMAN then? |
| 22:56.14 | bsdice | because its like calling a fax |
| 22:56.22 | bsdice | blind entry of pin |
| 22:56.23 | [TK]D-Fender | Prompt them for input with something else then. |
| 22:56.27 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com) |
| 22:57.07 | Kobaz | do de do |
| 22:57.17 | Kobaz | oh, you may know this |
| 22:57.28 | Kobaz | [TK]D-Fender: what are the possible authentication methods for sip? |
| 22:57.37 | Kobaz | there's like the digest response... |
| 22:57.55 | Kobaz | and then what others if any? I'm having trouble getting google to answer this one |
| 22:58.17 | [TK]D-Fender | Kobaz: I don't reallyknow the spec all that well. digest/realm is part, secret is another. caller vs authname, IP, etc |
| 23:00.41 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.174) |
| 23:03.21 | Kobaz | mm |
| 23:03.41 | Kobaz | basically my goal in this fishing is to see if there's any plaintext authentication that's part of the spec |
| 23:04.08 | Kobaz | [TK]D-Fender: a customer had a sip credential leek and international calls were made, but the carrier caught it quickly |
| 23:04.38 | [TK]D-Fender | Kobaz: IDS vs "Security" |
| 23:04.40 | Kobaz | we have strong passwords and all the usual security stuff turned on |
| 23:04.50 | Kobaz | [TK]D-Fender: true |
| 23:05.25 | Kobaz | no failed authenticated calls, the remote user just registered first shot and started banging away |
| 23:05.43 | Kobaz | i probably need to start implementing ids type stuff for asterisk |
| 23:05.57 | Kobaz | i got the rest of the system covered, with like tripwire and friends |
| 23:07.01 | [TK]D-Fender | check IP's, check area/country-codes, calls/hour, simultaneous channels, etc. |
| 23:10.58 | Kobaz | yeah |
| 23:11.18 | Kobaz | one problem that i noticed is the customer didn't set up the external access corretly |
| 23:11.39 | BarthezZ | you have about zero practical defense against those attacks :x |
| 23:11.40 | Kobaz | he's proxying the sip traffic, so the ast box only sees a static 192.168.x connecting in |
| 23:14.28 | Kobaz | the one thing i keep thinking about is how an attacker got a login... and i'm thinking like sniffing the air at a starbucks and caught a sip invite with plaintext credentials from a misconfigured client? |
| 23:14.40 | Kobaz | but i don't know if there is a plaintext auth for sip |
| 23:16.54 | BarthezZ | yeah there is... but I think allmost anything defaults to digest |
| 23:21.07 | BarthezZ | But... client side exploits are likely and plausible either for stealing credentials |
| 23:25.49 | Kobaz | a virused up smartphone perhaps |
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